Hey, thanks a lot for this plugin, I've been using it for a while - I like the output and configurability better than the PLII setting on my Z-5500s.

However, I think I've found a bug; take a look:

Notice how the FL, FR, C and RL outputs all look similar while the RR channel is almost the exact opposite of the RL channel. I think by some kind of glitch the phase of the RR channel is getting inverted. I'm in upmix mode, 2->6 channels and "Invert Phase" is turned off in the settings (though that shouldn't cause any problems like this either way) and it's not just an issue with this song, I've noticed it many times before.

Hello, and thanks for this plugin, rather than describing the settings I am currently using. Could you please tell me what the correct settings for a setup running foobar connected to an onkyo tx8211 stereo receiver, I have 6 speakers connected to the receiver, 2 large fisher speakers in front to channel "A" and 2 small and 2 medium speakers connected to channel B. These speakers are placed throughout my basement weight lifting room and I really don't know what settings on the plugin to use because I'm not sure if this is still a stereo setup or a 6 speaker setup for configuration purposes. Any help is greatly appreciated, I am also using the equalizer on foobar, the noise sharpening plugin, crossfader, and noise remover. It sounds pretty good already playing FLAC files

this is not a surround receiver, which is why I thought this plugin is necessary.

You don't need this plugin at all, although you could set it to "2 channels" . It's main use is to upmix from stereo to a 4.0 (4.1) or 5.1 setup (you would need some kind of surround set to be able to play that).

(1) If upmix mode is in "copy", why is there separate center and subwoofer volume levels if these are supposed to be just a raw left and right signals?

(2) Do delays work even when the source is six channel? Or is the whole plugin bypassed?

(3) I'm trying to upmix from stereo to 6 channels. What is the calculation for the subwoofer if you choose 'from all 5.1 sources'? I was hoping for a simple L+R calc. But if it forms the other 5 channels FIRST, to compute the sub channel, then it would be the sum of the following:

FR+FL+RR+RL+C

= R + L + (R-L)/2 + (L-R)/2 + (L+R)/2 (assuming FiR=0 and RiF=0)

= (3R + 3L)/2

Which means I'd have to knock the amplitude down by 1/3. Or is this normalization already done?

Interpolation would be too ponderous, with doubtful effect, so I had not considered its use at all. I prefer to resample all sources to 96kHz (PPHS) before any other processing (my sound card is 24/96 capable).

So what happens when I leave it at 44.1kHz and set the delay to 1ms? It rounds it off and delays it 44 samples (corresponding to 0.9977ms)?

I don't suppose there's any other benefit to upsampling, is there? I'm just using channel mixer for the delays and the L+R and L-R computations. I'm not using the filtering or any of that. I'd like to maintain signal purity if possible.

So what happens when I leave it at 44.1kHz and set the delay to 1ms? It rounds it off and delays it 44 samples (corresponding to 0.9977ms)?

Yes, it is.

QUOTE (StriatedFoot @ May 31 2010, 20:41)

I don't suppose there's any other benefit to upsampling, is there? I'm just using channel mixer for the delays and the L+R and L-R computations. I'm not using the filtering or any of that. I'd like to maintain signal purity if possible.

There is another benefit for older Creative (or other) cards, that do not have a native 44.1 support. Professional cards do not require software resampling for good sound quality.

Upon further testing, it appears there might be something wrong with this plugin.

In 6-channel mode, with upmix=surround, I expected the first two channels to be identical to the original two-channel waveform, as long as the RIF slider is set to zero. However, it is not. I can provide output files or screenshots of waveforms if needed.

Please let me know what the problem could be or if I'm doing something wrong. Like I said, I can provide you with raw waveforms, screenshots of waveforms, screenshots of settings, etc. Thank you.

The computation is pretty simple:OL = (IL-OBL*rif);OR = (IR-OBR*rif);so, if the RIF slider is set to zero, then output left is exactly indentical to input left. check your setup on another pages - subwoofer and delays.

The computation is pretty simple:OL = (IL-OBL*rif);OR = (IR-OBR*rif);so, if the RIF slider is set to zero, then output left is exactly indentical to input left. check your setup on another pages - subwoofer and delays.

I've uploaded three wav files (temporarily). The first, orig.wav, is the source wav. The second, cm.wav, is the output of channel mixer + matrix mixer (applied to remove the last four channels). The third, mm.wav, is the output of matrix mixer creating an L, R, L+R, L+R, L-R, and R-L channels.

From an audible standpoint, you should be able to hear the difference between orig.wav and cm.wav. The waveforms are also different visually, as shown below (left channel shown only).

I created these files by using foobar2000's wav converter. The settings for channel mixer were as follows:

I performed another test where I inserted channel mixer in between the two instances of matrix mixer that I used to create mm.wav. This time I left upmix = off but all the other settings the same (including time delay enabled but set to zero). It provided a bitperfect output. So the problem appears to be in the upmixing procedure itself.

I don't know too much about audio stuff, but I was just wondering what might be the best settings on these sliders to downmix 5.1 to 4 channel audio? I looked around to see if this question could be easily answered and promptly got confused by all of the settings =P

And also have it apply the changes so if I continue to hold the down key it will begin to increment the delay, apply, increment, apply...

The reason I ask is when setting up time alignment in a vehicle I typically use pink noise and work with 2 speakers at a time (starting at the furthest speaker). As part of using pink noise you can sweep up in delay, and down in delay till you find the rough area, then slowly increment to lock in the 'dopler effect'.

This method works well in a car environment because just raw measurements just don't take into account reflections.